Legal claims defining the scope of protection. Each claim is shown in both the original legal language and a plain English translation.
1. A method for making a call, comprising: detecting, by processing circuitry of a first terminal device that conducts a media call with a second terminal device via a network and in response to a success reception of a first data packet of first media that is send by the second terminal device during the media call, whether a packet loss occurs in the first media; obtaining, by the processing circuitry, network status information of the network when a packet loss of a second data packet of the first media is detected; determining, by the processing circuitry, a probability threshold in association with a retransmission request to the second terminal device for retransmitting the second data packet; determining, by the processing circuitry, based on the probability threshold, a network requirement condition for the retransmission request; sending, via interface circuitry of the first terminal device, the retransmission request to the second terminal device when the network status information satisfies the network requirement condition; and canceling the retransmission request to the second terminal device when the network status information fails to satisfy the network requirement condition.
2. The method according to claim 1 , further comprising: determining whether a first network state of the network that is indicated by the network status information matches a second network state that is required for retransmitting the second data packet; determining that the network status information satisfies the network requirement condition when the first network state matches the second network state; and determining that the network status information fails to satisfy the network requirement condition when the first network state does not match the second network state.
This invention relates to network communication systems, specifically methods for retransmitting data packets based on network state conditions. The problem addressed is ensuring reliable data transmission by evaluating network conditions before retransmitting packets, particularly in scenarios where network performance may fluctuate. The method involves analyzing network status information to determine whether current network conditions meet predefined requirements for retransmitting a data packet. This includes comparing a first network state, as indicated by the network status information, against a second network state that is required for successful retransmission. If the first network state matches the second required state, the network status information is deemed to satisfy the network requirement condition, allowing retransmission to proceed. Conversely, if the first network state does not match the required state, the network status information fails to satisfy the condition, and retransmission is either delayed or modified. The network status information may include metrics such as bandwidth, latency, packet loss, or other performance indicators. The second network state represents the optimal or acceptable conditions for retransmission, which may be predefined based on application requirements or dynamically adjusted. This approach ensures that retransmissions occur only when network conditions are favorable, improving efficiency and reliability in data transmission.
3. The method according to claim 2 , wherein the determining whether the first network state of the network that is indicated by the network status information matches the second network state that is required for retransmitting the second data packet comprises at least one of: determining whether a difference between a bandwidth threshold and a bandwidth in use is less than a first preset value; determining whether a transmission delay is less than a transmission delay threshold; determining whether a packet loss rate is less than a packet loss rate threshold; and determining whether a number of consecutive lost packets is less than a second preset value.
This invention relates to network communication systems, specifically methods for determining whether a network state is suitable for retransmitting data packets. The problem addressed is ensuring reliable data transmission by evaluating network conditions before retransmitting lost or corrupted packets. The method involves analyzing network status information to assess whether the current network state meets predefined criteria for retransmission. The method compares the current network state against required conditions, which include evaluating bandwidth availability, transmission delay, packet loss rate, and consecutive lost packets. Specifically, it checks if the difference between a bandwidth threshold and the current bandwidth in use is below a first preset value, ensuring sufficient available bandwidth. It also verifies if the transmission delay is below a threshold, the packet loss rate is below a threshold, and the number of consecutive lost packets is below a second preset value. These checks collectively determine whether the network conditions are favorable for retransmission, improving data transmission reliability by avoiding retransmissions under poor network conditions. The method ensures efficient use of network resources while minimizing packet loss and delays.
4. The method according to claim 1 , further comprising: analyzing, by the processing circuitry, a signal feature of a media segment in the first data packet; and sending, via the interface circuitry, the retransmission request when the network status information satisfies the network requirement condition and the signal feature is indicative of a semantic importance.
This invention relates to a method for optimizing retransmission requests in a communication system, particularly for media data transmission where network conditions and semantic importance of data segments influence retransmission decisions. The method involves monitoring network status information, such as latency, bandwidth, or packet loss, to determine if it meets predefined network requirement conditions. If the network status is favorable, the system analyzes a signal feature of a media segment within a received data packet to assess its semantic importance. Semantic importance may be determined by factors such as audio loudness, video motion complexity, or other content-based metrics. If the network status meets the requirements and the media segment is deemed semantically important, a retransmission request is sent to ensure high-priority data is reliably delivered. This approach prioritizes retransmission for critical media segments while avoiding unnecessary retransmissions for less important data, improving efficiency and quality of service in media streaming or real-time communication applications. The method may be implemented in processing circuitry with interface capabilities to handle data packets and network status monitoring.
5. The method according to claim 1 , further comprising: extracting, by the processing circuitry, a sequence number index in the first data packet; and determining whether the packet loss occurs in the first media based on the sequence number index.
This invention relates to detecting packet loss in media data transmission, particularly in systems where data packets are sequentially ordered. The problem addressed is the need for efficient and accurate detection of lost packets in a media stream, which is critical for maintaining data integrity and quality in real-time communication or streaming applications. The method involves processing a first data packet of a media stream to detect packet loss. The processing circuitry extracts a sequence number index from the first data packet, which is a unique identifier assigned to each packet in the sequence. By analyzing this index, the system determines whether a packet loss has occurred. This is done by comparing the sequence number index of the first data packet with expected values or previous indices to identify missing packets in the sequence. The method ensures reliable detection of packet loss, enabling timely corrective actions such as retransmission or error concealment. The invention may also include additional steps such as analyzing packet headers or payloads to further verify packet loss. The system may use the sequence number index in conjunction with other metadata to improve accuracy. This approach is particularly useful in applications like video conferencing, live streaming, or any system where uninterrupted media transmission is essential. The method enhances data reliability and user experience by minimizing disruptions caused by lost packets.
6. The method according to claim 1 , further comprising at least one of: determining a first network requirement condition to ensure that a first probability for the retransmission request to arrive at the second terminal device within a buffer time is equal to or higher than the probability threshold, the first media being allowed to be buffered at the second terminal device in the buffer time; or determining a second network requirement condition to ensure that a second probability to output the second data packet at the second terminal device is equal to or higher than the probability threshold.
This invention relates to optimizing data transmission in a network, particularly for ensuring reliable delivery of media data between terminal devices. The problem addressed is the potential for data packet loss or delay during transmission, which can disrupt media playback or other time-sensitive applications. The solution involves dynamically adjusting network conditions to improve the likelihood of successful data delivery. The method includes determining network requirement conditions to enhance the probability of meeting specific performance thresholds. One condition ensures that a retransmission request for a first media data packet will arrive at a receiving terminal device within a specified buffer time, allowing the media to be buffered without interruption. Another condition ensures that the probability of outputting a second data packet at the receiving terminal is sufficiently high, maintaining data integrity and timely delivery. These conditions are derived based on network metrics such as latency, bandwidth, and packet loss rates, and may involve adjusting transmission parameters like retransmission timers or buffer sizes. The approach dynamically adapts to network fluctuations to minimize disruptions in media streaming or other real-time applications.
7. The method according to claim 1 , further comprising: collecting offline network data; extracting at least one network parameter for representing a network feature from the offline network data; constructing a network model based on the at least one network parameter; determining a first de-jittering policy based on the network model; modifying, by the processing circuitry, the first de-jittering policy based on a feature parameter for evaluating a call quality of the media call to obtain a second de-jittering policy; obtaining a de-jittering parameter based on the network status information and the second de-jittering policy; and setting, at the first terminal device, a capacity of a buffer that is used to for buffering transmission data during the media call based on the de-jitter parameter to ensure a delay of the media call meet an expectation.
This invention relates to optimizing media call quality in network communications by dynamically adjusting de-jitter buffering. The problem addressed is maintaining smooth media transmission despite network variability, which can cause jitter and delay. The solution involves analyzing offline network data to extract parameters representing network features, then constructing a network model to determine an initial de-jittering policy. This policy is refined based on call quality metrics to produce a second, optimized policy. Using real-time network status information and the refined policy, a de-jittering parameter is derived to dynamically adjust the buffer capacity of a terminal device during a media call. This ensures the call delay meets quality expectations by balancing buffering against latency. The approach leverages historical and real-time network data to adaptively manage buffering, improving media call performance in variable network conditions.
8. The method according to claim 7 , wherein the modifying the first de-jittering policy based on the feature parameter for evaluating the call quality of the media call, to obtain the second de-jittering policy comprises: obtaining at least one of a signal content and an auditory perception result of the media call; and modifying the first de-jittering policy based on the at least one of the signal content and the auditory perception result of the media call.
This invention relates to adaptive de-jittering techniques for improving media call quality in communication systems. Jitter, or variability in packet arrival times, can degrade audio or video quality in real-time media calls. Traditional de-jittering methods use fixed policies, which may not optimize quality for varying call conditions. The invention describes a method for dynamically adjusting de-jittering policies based on call-specific features. It involves analyzing the media call to obtain signal content (e.g., speech, music, or noise characteristics) and auditory perception results (e.g., perceived quality metrics). These features are used to modify an initial de-jittering policy, generating a second, optimized policy tailored to the call's specific conditions. For example, speech calls may prioritize low latency, while music calls may prioritize smooth playback. The method ensures that de-jittering adapts to real-time changes in call quality, enhancing user experience. The invention builds on a prior step of determining a feature parameter for evaluating call quality, which may include metrics like packet loss, delay, or signal-to-noise ratio. By incorporating signal content and auditory perception, the method refines the de-jittering approach beyond traditional statistical methods, addressing the problem of static policies failing to adapt to diverse media types and network conditions. The result is a more responsive and context-aware de-jittering system.
9. The method according to claim 1 , further comprising: obtaining, by the processing circuitry of the first terminal device, based on the first media, a far-end signal that is sent by the second terminal device during the media call; superimposing, by the processing circuitry, an ultrasonic signal on the far-end signal to obtain a mixed signal; playing, via a speaker, the mixed signal; obtaining, by the processing circuitry of the first terminal device, a near-end signal that is generated by a microphone; determining a first signal segment in the mixed signal and a second signal segment in the near-end signal based on the ultrasonic signal; calculating a correlation value between the first signal segment and the second signal segment; and determining that the media call is in a both-speaking state when the correlation value is less than a preset correlation value threshold.
This invention relates to real-time communication systems, specifically detecting when two participants in a media call (e.g., a voice or video call) are speaking simultaneously, a condition known as a "both-speaking state." The problem addressed is accurately identifying this state to improve call quality, reduce echo, or enhance noise suppression. The method involves a first terminal device (e.g., a smartphone or computer) processing signals during a media call with a second terminal device. The device obtains a far-end signal (audio from the second terminal) and superimposes an ultrasonic signal onto it, creating a mixed signal. This mixed signal is played through the speaker. The device also captures a near-end signal (audio from its own microphone). By analyzing the ultrasonic signal, the device identifies corresponding segments in the mixed and near-end signals. A correlation value is calculated between these segments. If the correlation value falls below a preset threshold, the system determines that both participants are speaking simultaneously. This detection can trigger actions like adjusting audio processing or notifying users to avoid overlapping speech. The ultrasonic signal serves as a reference to distinguish the far-end signal from the near-end signal, enabling precise correlation analysis. The method improves upon traditional echo cancellation techniques by leveraging ultrasonic markers for more accurate both-speaking detection.
10. The method according to claim 9 , further comprising: superimposing the ultrasonic signal that is encoded with data on the far-end signal to obtain the mixed signal; determining, in the near-end signal, the second signal segment that carries specific data in a frequency range corresponding to the ultrasonic signal; and determining, in the mixed signal, the first signal segment that is superimposed with the ultrasonic signal that is encoded with the specific data.
A method and system for data communication in a device, particularly for applications such as ultrasonic communication or covert data transmission. The technology addresses the challenge of embedding and recovering data within existing communication signals. The process involves transmitting a signal from a far-end device. Simultaneously, an ultrasonic signal, which is encoded with specific data, is superimposed onto this far-end signal to create a mixed signal. The near-end device receives this mixed signal. To extract the encoded data, the near-end device first analyzes the received signal to identify a segment within the near-end signal that corresponds to the ultrasonic signal's frequency range and carries specific data. This identified segment is then used to locate a corresponding segment within the mixed signal. This specific segment in the mixed signal is identified as the one that has the ultrasonic signal superimposed with the specific data. This allows for the selective detection and decoding of the embedded ultrasonic data from the combined signal.
11. An apparatus, comprising: interface circuitry configured to transmit and receive signals carrying media data to/from a network during a media call with another apparatus; and processing circuitry configured to: detect, in response to a success reception of a first data packet of first media that is send by the other apparatus during the media call, whether a packet loss occurs in the first media; obtain network status information of the network when a packet loss of a second data packet of the first media is detected; determine a probability threshold in association with a retransmission request to the other apparatus for retransmitting the second data packet; determine based on the probability threshold, a network requirement condition for the retransmission request; send, via the interface circuitry, the retransmission request to the other apparatus when the network status information satisfies the network requirement condition; and cancel the retransmission request to the other apparatus when the network status information fails to satisfy the network requirement condition.
Communication network systems and methods for efficient media data transmission. The invention addresses the problem of managing retransmission requests for lost media data packets during a media call to optimize network resource utilization and maintain call quality. The apparatus includes interface circuitry for transmitting and receiving media data signals to and from a network during a media call. Processing circuitry is configured to detect packet loss in received first media data. Upon detecting packet loss for a specific data packet of the first media, the processing circuitry obtains network status information. It then determines a probability threshold associated with a retransmission request for the lost data packet. Based on this probability threshold and the obtained network status information, the processing circuitry determines if a network requirement condition for sending the retransmission request is met. If the network status information satisfies this condition, the retransmission request is sent via the interface circuitry to the other apparatus involved in the media call. Conversely, if the network status information fails to satisfy the network requirement condition, the retransmission request is canceled.
12. The apparatus according to claim 11 , wherein the processing circuitry is configured to: determine whether a first network state of the network that is indicated by the network status information matches a second network state that is required for retransmitting the second data packet; determine that the network status information satisfies the network requirement condition when the first network state matches the second network state; and determine that the network status information fails to satisfy the network requirement condition when the first network state does not match the second network state.
In the field of network communication, ensuring reliable data transmission is critical, especially in dynamic network environments where conditions may fluctuate. The invention addresses the challenge of determining whether current network conditions are suitable for retransmitting data packets, particularly when retransmission is necessary due to packet loss or corruption. The apparatus includes processing circuitry that evaluates network status information to assess whether the network's current state meets predefined requirements for retransmitting a data packet. The processing circuitry compares the first network state, as indicated by the network status information, against a second network state that is required for successful retransmission. If the first network state matches the second required state, the network status information is deemed to satisfy the network requirement condition, enabling retransmission. Conversely, if the first network state does not match the required state, the network status information fails to meet the condition, indicating that retransmission may not be feasible or optimal under current conditions. This mechanism ensures that retransmissions occur only when network conditions are favorable, improving efficiency and reliability in data communication.
13. The apparatus according to claim 12 , wherein the processing circuitry is configured to determine at least one of: whether a difference between a bandwidth threshold and a bandwidth in use is less than a first preset value; whether a transmission delay is less than a transmission delay threshold; whether a packet loss rate is less than a packet loss rate threshold; and whether a number of consecutive lost packets is less than a second preset value.
This invention relates to network communication systems, specifically to apparatuses that monitor and optimize network performance by evaluating multiple quality metrics. The problem addressed is the need for efficient and reliable network performance assessment to ensure optimal data transmission. The apparatus includes processing circuitry that analyzes network conditions by determining whether specific performance metrics meet predefined thresholds. These metrics include the difference between a bandwidth threshold and the current bandwidth in use, ensuring sufficient available bandwidth. The system also checks if the transmission delay is below a specified threshold, indicating acceptable latency. Additionally, it evaluates the packet loss rate to ensure it remains below a threshold, and monitors the number of consecutive lost packets to detect potential network instability. By assessing these factors, the apparatus can dynamically adjust network operations to maintain high-quality communication. The processing circuitry's configuration allows for real-time monitoring and decision-making based on multiple criteria, improving network reliability and efficiency. This approach helps prevent performance degradation by proactively identifying and addressing potential issues before they impact data transmission. The system is particularly useful in environments where consistent and high-quality network performance is critical, such as in real-time applications or large-scale data transfers.
14. The apparatus according to claim 11 , wherein the processing circuitry is configured to: analyze a signal feature of a media segment in the first data packet; and send, via the interface circuitry, the retransmission request when the network status information satisfies the network requirement condition and the signal feature is indicative of a semantic importance.
This invention relates to network communication systems, specifically improving retransmission decisions for media data packets based on both network conditions and semantic importance of the content. The problem addressed is inefficient retransmission of media data, which can waste bandwidth on unimportant content or fail to retransmit critical content, degrading user experience. The apparatus includes processing circuitry and interface circuitry for network communication. The processing circuitry analyzes network status information to determine if it meets a predefined network requirement condition, such as available bandwidth or latency thresholds. Additionally, the processing circuitry evaluates a signal feature of a media segment within a data packet to assess its semantic importance, such as detecting key frames, high-priority audio segments, or other content with significant perceptual or informational value. If both the network condition is satisfied and the signal feature indicates semantic importance, the apparatus sends a retransmission request via the interface circuitry. This ensures that retransmissions prioritize critical content when network conditions allow, optimizing bandwidth usage and improving media quality. The system may also include a memory for storing network status information and media data, and a display for user interaction. The retransmission request may be sent to a network node or another device in the communication system.
15. The apparatus according to claim 11 , wherein the processing circuitry is configured to: determine a first network requirement condition to ensure that a first probability for the retransmission request to arrive at the other apparatus within a buffer time is equal to or higher than the probability threshold, the first media being allowed to be buffered at the other apparatus in the buffer time; or determine a second network requirement condition to ensure that a second probability to output the second data packet at the other apparatus is equal to or higher than the probability threshold.
This invention relates to network communication systems, specifically addressing the challenge of ensuring reliable data transmission with probabilistic guarantees. The apparatus includes processing circuitry that manages data packet transmission between devices, focusing on minimizing retransmission delays and ensuring timely data delivery. The processing circuitry determines network conditions to meet specific probability thresholds for retransmission requests or data output. For retransmission requests, it ensures a first probability that the request arrives at the receiving device within a predefined buffer time is at least a set threshold, allowing the first media (e.g., data packets) to be buffered without loss. Alternatively, it ensures a second probability that the second data packet is output at the receiving device meets or exceeds the threshold, guaranteeing timely delivery. The apparatus may also include a transmitter for sending data packets and a receiver for receiving acknowledgments or retransmission requests. The processing circuitry adjusts transmission parameters based on network conditions, such as latency, packet loss, or bandwidth, to optimize reliability. This ensures that media data, such as video or audio streams, is delivered with minimal delay and high probability of successful transmission, addressing issues in real-time communication systems where timing and reliability are critical.
16. The apparatus according to claim 11 , wherein the processing circuitry is configured to: collect offline network data; extract at least one network parameter for representing a network feature from the offline network data; construct a network model based on the at least one network parameter; determine a first de-jittering policy based on the network model; modify the first de-jittering policy based on a feature parameter for evaluating a call quality of the media call to obtain a second de-jittering policy; obtain a de-jittering parameter based on the network status information and the second de-jittering policy; and set a capacity of a buffer that is used to for buffering transmission data during the media call based on the de-jitter parameter to ensure a delay of the media call meet an expectation.
This invention relates to network communication systems, specifically addressing the challenge of managing jitter in media calls to ensure smooth transmission and acceptable call quality. Jitter, or variability in packet arrival times, can degrade real-time media streams such as voice or video calls. The apparatus described processes offline network data to extract network parameters that represent network features, such as latency, packet loss, or bandwidth fluctuations. These parameters are used to construct a network model that predicts network behavior. Based on this model, a de-jittering policy is determined to mitigate jitter effects. The policy is then refined using a feature parameter that evaluates call quality, resulting in an optimized de-jittering policy. The apparatus further obtains a de-jittering parameter by analyzing real-time network status information and applying the refined policy. Finally, the buffer capacity for transmitting media data during the call is adjusted based on this de-jittering parameter to ensure the call delay meets expected performance standards. This approach dynamically adapts to network conditions to maintain high-quality media transmission.
17. The apparatus according to claim 16 , wherein the processing circuitry is configured to: obtain at least one of a signal content and an auditory perception result of the media call; and modify the first de-jittering policy based on the at least one of the signal content and the auditory perception result of the media call.
This invention relates to improving media call quality by dynamically adjusting de-jittering policies in communication systems. The problem addressed is the static nature of traditional de-jittering algorithms, which fail to adapt to varying call conditions, leading to suboptimal audio quality. The apparatus includes processing circuitry that monitors media calls to assess signal content and auditory perception results. Based on these evaluations, the circuitry modifies the de-jittering policy in real-time. Signal content analysis may involve detecting speech patterns, background noise, or packet loss, while auditory perception results could include user feedback or objective quality metrics. The dynamic adjustment ensures that de-jittering parameters, such as buffer size or delay thresholds, are optimized for the current call conditions, enhancing clarity and reducing artifacts. This adaptive approach improves user experience by mitigating issues like echo, distortion, or latency that arise from fixed de-jittering settings. The invention is particularly useful in VoIP, video conferencing, and real-time communication systems where call quality varies due to network fluctuations or content characteristics.
18. The apparatus according to claim 11 , wherein the processing circuitry is configured to: obtain, based on the first media, a far-end signal that is sent by the other apparatus during the media call; superimpose an ultrasonic signal on the far-end signal to obtain a mixed signal; play, via a speaker, the mixed signal; obtain a near-end signal that is generated by a microphone; determine a first signal segment in the mixed signal and a second signal segment in the near-end signal based on the ultrasonic signal; calculate a correlation value between the first signal segment and the second signal segment; and determine that the media call is in a both-speaking state when the correlation value is less than a preset correlation value threshold.
This invention relates to media call systems, specifically detecting when both parties are speaking simultaneously (a "both-speaking state") during a call. The problem addressed is accurately identifying such states to improve call quality, reduce echo, or enhance noise suppression. The apparatus includes processing circuitry that obtains a far-end signal from another device during a call. An ultrasonic signal is superimposed onto this far-end signal to create a mixed signal, which is then played through a speaker. A microphone captures a near-end signal, which includes the user's voice and the mixed signal. The processing circuitry identifies segments of the mixed signal and near-end signal that contain the ultrasonic signal. It then calculates a correlation value between these segments. If the correlation value falls below a preset threshold, the system determines that both parties are speaking simultaneously. This method leverages the ultrasonic signal as a reference to distinguish between the far-end and near-end audio components, enabling precise detection of overlapping speech. The invention improves upon prior art by using an ultrasonic marker to enhance the accuracy of both-speaking state detection, which is crucial for real-time audio processing in telecommunication systems.
19. The apparatus according to claim 18 , wherein the processing circuitry is configured to: superimpose the ultrasonic signal that is encoded with data on the far-end signal to obtain the mixed signal; determine, in the near-end signal, the second signal segment that carries specific data in a frequency range corresponding to the ultrasonic signal; and determine, in the mixed signal, the first signal segment that is superimposed with the ultrasonic signal that is encoded with the specific data.
This invention relates to an apparatus for processing audio signals to enable data transmission via ultrasonic signals superimposed on audio communications. The problem addressed is the need to transmit data covertly or efficiently within existing audio communication channels without disrupting the primary audio signal. The apparatus includes processing circuitry configured to superimpose an ultrasonic signal, encoded with data, onto a far-end signal to produce a mixed signal. The far-end signal is typically the audio signal received from a remote source in a communication system. The processing circuitry then analyzes the near-end signal, which is the audio signal captured by a local microphone, to identify a second signal segment containing specific data within a frequency range corresponding to the ultrasonic signal. Additionally, the processing circuitry identifies a first signal segment in the mixed signal where the ultrasonic signal, carrying the specific data, is superimposed. This allows for the extraction and processing of the embedded data while preserving the integrity of the primary audio communication. The apparatus ensures that data transmission occurs in a frequency range that does not interfere with the audible audio signal, enabling seamless integration of data communication within standard audio systems. This method is particularly useful in applications requiring covert data transfer or efficient use of existing communication channels.
20. A non-transitory computer-readable medium storing instructions which when executed by a computer cause the computer to perform: detecting, by the computer that conducts a media call with another computer via a network and in response to a success reception of a first data packet of first media that is send by the other computer during the media call, whether a packet loss occurs in the first media; obtaining network status information of the network when a packet loss of a second data packet of the first media is detected; determining a probability threshold in association with a retransmission request to the other computer for retransmitting the second data packet; determining based on the probability threshold, a network requirement condition for the retransmission request; sending the retransmission request to the other computer when the network status information satisfies the network requirement condition; and canceling the retransmission request to the other computer when the network status information fails to satisfy the network requirement condition.
A computer system and method for managing media calls over a network, specifically addressing packet loss in transmitted media. The system aims to optimize retransmission requests to avoid unnecessary network traffic. The invention involves a non-transitory computer-readable medium storing instructions. When executed, these instructions enable a computer, engaged in a media call with another computer via a network, to detect packet loss for first media received from the other computer. Upon detecting packet loss for a subsequent data packet of the same media, the system obtains network status information. It then determines a probability threshold related to retransmission requests and establishes a network requirement condition for sending such a request based on this threshold. A retransmission request for the lost packet is sent to the other computer only if the obtained network status information meets this condition. If the network status information does not satisfy the condition, the retransmission request is canceled.
Unknown
June 23, 2020
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